This is difficult to encounter in ordinary networks, but is
encountered when using tc-netem to add random delays to packets, and
also when your UDP stream is bonded over multiple links with varying
characteristics.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1979>
This patch fixes a seg.fault in gst_structure_new() with warnings as below.
GLib-GObject-WARNING **:
../gobject/gtype.c:4330: type id '0' is invalid
GLib-GObject-WARNING **:
can't peek value table for type '<invalid>' which is not currently referenced
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1920>
When doing only a single stream of audio/video this hardly matters,
but when doing many at the same time, the fact that you have to get
a hold of the glib global type-system lock every time you process a buffer,
means that there is a limit to how many streams you can process in
parallel.
Luckily the fix is very simple, by doing a cast rather than a full
type-check.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1890>
Not having this field is equivalent with it being 1/1 so consider
it like that. The generic caps functions are not aware of these
semantics and would consider the caps different, causing a negotiation
failure when caps are changing from caps with to caps without or the
other way around.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1833>
Not having these fields is equivalent with them being mono/0 so consider
them like that. The generic caps functions are not aware of these
semantics and would consider the caps different, causing a negotiation
failure when caps are changing from caps with to caps without or the
other way around.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1833>
LOAD macro relies in m7 being zero for interleaving purposes. Using LOAD
on the m7 register makes it interleave with its new content instead of
with 0.
The effect of this bug was bobbing on some static lines that appeared
over fast-moving content.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1823>
The RTP payload seems to be required as it carries the frame count
information. Also, gst_rtp_base_payload_allocate_output_buffer had
the second argument incorrect.
Strangely some devices like Shanling MP4 and Sony XM3 would still
work without this while some like the Sony XM4 do not.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1804>
When multiple streams are bundled together, there may be more
than one red payload type to handle.
In addition, as the red decoder works by filling in gaps in
the seqnums, there needs to be one rtp_history queue per sequence
domain.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1429>
In redenc, when input buffers have a header for the TWCC extension,
we now add one to our wrapper buffers.
In ulpfecenc we add one in that case to our protection buffers.
This makes TWCC functional when UlpRed is used in webrtcbin.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1414>
In order to allow "level-asymmetry-allowed" we now handle a new
"profile" field, which as the same semantics as the "profile" field in
H.264 stream so that we can force payloaded stream to have the right
format when using the `gst_sdp_media_get_caps_from_media` to set caps
filter after the payloader. This allows a simple negotiation in standard
RTP negotiation based on SDPs (like webrtc) for that particular case,
closely respecting the specs.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1410>
since `gst_caps_replace()` and `gst_pad_set_caps()` both ref the caps and neither of them takes the ownership of the caps -> it must be unreffed in `gst_multi_file_src_set_property()`
to test the leak (on Unix): `echo coucou > /tmp/file.txt && GST_TRACERS=leaks GST_DEBUG="GST_TRACER:7" gst-launch-1.0 multifilesrc location=/tmp/file.txt caps='txt' ! fakesink`
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1436>
Follow-up on 97d83056b3, only check
for intersection with the current srccaps when checking if a sinkpad
can accept caps.
I must have been lucky in my firefox testing then, and always entered
the code path with audio getting negotiated first, thus not failing
the is_subset check when srccaps had been negotiated as
application/x-rtp, and an accept-caps query was made for the video
caps with a defined extmap.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1384>
A previous patch has caused rtpfunnel to output twcc-related
information downstream, however this leaked into upstream
negotiation (through funnel->srccaps), causing payloader to
negotiate twcc caps even when not prompted to do so by the user.
Fix this by only enforcing that upstream sends us application/x-rtp
caps as was the case originally.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1278>
